sr-users

sr-users@lists.kamailio.org

42 participants
37455 discussions

[Kamailio-Users] ERROR:mysql:db_mysql_new_connection: driver error: Too many connections
by Samael - 04 Sep '08

04 Sep '08

Hi, We are deploying OpenSER in a large infrastructure together with Asterisk to support thousands of current transactions. We are using udp listener and mysql module in openser. We had to increase the children parameter of the openser.cfg to 200 because Openser couldn't handle the high amount of concurrent SIP REQUESTs from ASTERISK (Not only INVITES, but also a lot of SIP OPTIONs), that means that Openser couldn't handle more than 'children' REQUEST loosing the rest of the concurrent REQUEST. children=200 But then, the following error appeared in the openser: ERROR:mysql:db_mysql_new_connection: driver error: Too many connections This is because the Standard parameter of mysql "max_connections" is set to 100. In mysql CLI, we can see that every child process of openser has a mysql connection. (see in Mysql> SHOW PROCESSLIST) The recommended max_connections parameter is 500-1000 => we have a limit of 500-1000 concurrent SIP REQUEST to be processed by the openser. Why there is a process for every child? Why is neccesary a mysql connection for every child? Can the openser have a pool of mysql connection to be used by thousands of children? Regards

4 6

[Kamailio-Users] OpenSER reply 513 when register use FQDN
by Steven Wu 04 Sep '08

04 Sep '08

Hi, I just want use FQDN in my networking environment. My SJPhone use FQDN register to OpenSER 1.2.2 but was rejected by 513 Message to big. However it can be registed use the same profile just replace FQDN to IP address. The DNS query process worked fine and got the right IP address. My environment elments looks like: OpenSER 1.2.2 at host 192.168.1.120 / ipcsip.org DNS server at host 10.57.39.53 SJPhone at host 192.168.1.105 REGISTER package looks like: 192.168.1.105 -> 192.168.1.120 REGISTER sip:ipcsip.org:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062> Contact: <sip:1000@192.168.1.105> Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 192.168.1.120 -> 192.168.1.105 SIP/2.0 513 Message too big Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport=5060 From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.b7d8 Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 I got same result evenif enabled dns & rev_dns in openser.cfg (dns=yes rev_dns=yes). So, What wrong with me? Coudle some give me some instructions? Best regards, Steven Wu

4 7

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 04 Sep '08

04 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Serusers] Which kind of SIP message can I use to exchange Data out of session?
by Peter Cai 04 Sep '08

04 Sep '08

Hi All, If a SIP client has to exchange some data out of a Session with the proxy server, such as get his "buddy list", what SIP Message can I use? I didn't find any thing suit for this purpose. Should I use something beside SIP such as HTTP.

3 3

[Serusers] cannot load module acc.so
by 静于 04 Sep '08

04 Sep '08

hello, I cannot load the acc.so module.The config file is below: # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=10.1.2.3 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=bigu.edu alias=10.1.2.3 # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5060 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. uid="nobody" # group | gid - gid to be used by the server. 99 = nobody. gid="nobody" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external module loading loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ?�?;lr=true?�? modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:>password>@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:>password>@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(10\.1\.2\.3|(proxy\.)?bigu\.edu)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("bigu.edu", "subscriber")) { www_challenge("bigu.edu", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(10\.1\.2\.3|(proxy\.)?bigu\.edu)([:;\?].*)?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { route(3); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@10\.1\.2\.5([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==10.1.2.3) & !(proxy_authorize("bigu.edu", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("bigu.edu", "0"); break; }; }; # to maintain credibility of our proxy, we check >From in INVITEs if (!src_ip==10.1.2.3 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(10.1.2.3|(proxy\.)?bigu\.edu)") { if (!(src_ip==10.1.2.3 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("bigu.edu", "subscriber")) { proxy_challenge( "bigu.edu","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("10.1.2.5:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } Can anyone help BR Ivy ___________________________________________________________ �Ż��䣬��䣡 http://cn.mail.yahoo.com/

1 0

Re: [Kamailio-Users] svn diff
by Vadim Berezniker 04 Sep '08

04 Sep '08

Oops hit Ctrl+Enter svn diff http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/ http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.3/ I've been trying to run this against the public SVN but it never completes.... I hate sourceforge. From: Vadim Berezniker Sent: Wednesday, September 03, 2008 1:57 PM To: 'users(a)lists.kamailio.org' Subject: svn diff Can someone with access to the dev svn run this for me?

3 3

[Kamailio-Users] Could you please help me how to integrate RLS AS and PS AS with SERIMS?
by mahesh.peddi＠info-spectrum.com 04 Sep '08

04 Sep '08

Hi, I am working on openser 1.3.2 as RLS application server and presence application server. I want to integrate this two application servers with SERIMS. Could you please help me how to integrate and make run this. To do this I have run SERIMS components on two servers? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Kristian Kielhofner 03 Sep '08

03 Sep '08

Hello everyone, Feel free to set $OSSPROXY to whatever you like = Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). $OSSPROXY is my favorite SIP toolkit. In addition to being an excellent proxy and registrar it serves as a platform to do SIP mangling and wrangling. With that in mind let me propose a crazy idea... How hard would it be to get $OSSPROXY to be able to parse what is commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has no business getting too involved in the SDP but I feel it makes sense in some instances (like this one). What if I want to make a routing decision based on an encapsulated ISUP parameter of some sort? Who knows what it could be!?!? Maybe there are some other uses but that makes the most sense and seems to be the most practical. It should be possible to develop an $OSSPROXY module to: - Use the SDP parser to find the ISUP SDP part: http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing - Pass that part into a "real" ISUP parser (if necessary). I couldn't seem to find the perfect library but perhaps something like this could work: http://www.openss7.org/isup.html - Make the ISUP parameters available via AVPs, pseudo variables, etc for use in the $OSSPROXY script. I don't really have a need for this but it seems like it could be cool. Am I nuts? -- Kristian Kielhofner http://blog.krisk.org

3 3

[Serusers] Is setdsturi supported in SER
by Steven C. Blair 03 Sep '08

03 Sep '08

I am wondering if setdsturi is supported in SER 0.9.7-pre3 Thanks

1 0

[Kamailio-Users] svn diff
by Vadim Berezniker 03 Sep '08

03 Sep '08

Can someone with access to the dev svn run this for me?

1 0

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users